xe2x80x9cA Wireless Infrared Digital Audio Transmitting System,xe2x80x9d Ser. No.: 09/425,314, Filing Date: Oct. 25, 1999, assigned to the Same Assignee as the present invention.
xe2x80x9cA Wireless Infrared Digital Audio Receiving System,xe2x80x9d Ser. No.: 09/427,020, Filing Date: Oct. 25, 1999, assigned to the Same Assignee as the present invention.
1. Field of the Invention
This invention is generally related to electronic circuits and systems that transmit and receive digitally sampled analog signals. More particularly, this invention relates to electronic circuits and systems that transmit and receive digital audio signals.
2. Description of the Related Art
The transmission of analog signals between a source of the analog signals and a reproduction of the analog signals at an output transducer is well known in the art. U.S. Pat. No. 5,596,648 (Fast) describes a wireless infrared audio transmission system where infrared LED emitters are activated by a frequency modulated pulse wave transmitted as light to a receiver. The audio analog signal modulates the frequency modulated pulse wave.
U.S. Pat. No. 5,596,603 (Haupt, et al.) illustrates another device for wireless transmission of audio signals. Refer now to FIG. 1 for an overview of this structure. The analog source 5 provides a left channel L and a right channel R. The analog source 5 would be microphones, a FM tuner/receiver, or an analog recording media. The left channel L and the right channel R are inputs to analog-to-digital converters 15 and 20. It is well known in the art that the analog sources can provide any number of channels. The left channel L and right channel R are chosen for illustration purposes.
Additionally, the analog signals from the analog source 5 can have been previously converted to digitized samples and then provided by the digital source 10. The digitized samples of the analog signals are retained in a data buffer 25. The digitized samples are then formatted in data frames in the data formatting unit 30. In Haupt, et al. a data frame is 128 bits in length for each channel (left channel L or right channel R). The data frames are then transferred to the date modulator 35. A carrier signal is then modulated with the data frames.
In the case of Haupt, et al., the data frames are changed from a 4 bit audio data to a 5 bit transmission data, which is used to activate and deactivate an infrared light emitting diode. The modulated carrier signal is transferred to a transmitter and then conveyed to the communication medium 45. The infrared light is then radiated through the open atmosphere to a receiving light sensitive diode. In this case, the communication medium 45 is the open atmosphere.
It is well known that the transmitter 40 can produce radio frequency waves in addition to light. Further, the communication medium 45 can be either wire such as coaxial cable, twisted-pair cable or other forms of metallic (copper) inter-connection. Additionally, the communication medium 45 may be a fiber optic cable.
The receiver 50 will recover the modulated carrier signal from the communication medium 45. Typically, a clocking or timing signal is included in the data frame and the modulated carrier signal. A clock extraction circuit 55 will develop the embedded clocking or timing signal and synchronize the receiving subsystem 100 with transmitting subsystem 95. Classically, the clock extraction circuit 55 incorporates a phase locked oscillator, which can malfunction if there are errors in the transmitted modulated carrier signal.
The recovered modulated carrier signal is transferred to the demodulator 60 to extract the data frames. The data frames are then reformatted in the receive data formatter 65 to recreate the digitized samples of the analog signals. The recreated digitized samples are then transferred to the digital-to-analog converters 70 and 75 to reproduce the analog signals 80 and 85. Alternately, the digitized samples of the analog data can be transferred 95 to external circuitry for further processing.
The wireless transmission as shown in FIG. 1 is subject to corruption of the digitized samples during transmission. For instance, noise from an electronically ballasted halogen lamp would completely breakdown recovery of the transmission of the modulated carrier signal.
A solution to the corruption of the modulated carrier signal is to provide a level of redundancy for the digitized samples. U.S. Pat. No. 5,832,024 (Schotz, et al.) shows the use of forward error correction codes using the well known Reed-Solomon Coding. This will eliminate errors of relatively short duration, but will not prevent disruption of the output analog signals 80 and 85 due to long term digitized sample corruption.
To eliminate longer corruption of the digitized samples Schotz, et al. employs a convolutional interleaving circuit to separate the digitized samples of the analog signal that would normally be transmitted together. This allows the greater probability that a longer term error can be to be corrected
If the error correction coding and the convolutional interleaving of the digitized samples cannot insure corrected digitized samples of the analog signals, the analog signal will be reproduced (especially in audio signals) as annoying cracks and pops in a speaker. To eliminate the cracks and pops, Schotz, et al. suggests that the digitized samples can be brought to a null level or muted. However, if the muting is activated suddenly, it is distracting and is annoying to the listener in an audio application.
U.S. Pat. No. 5,602,669 (Chaki) provides a digital transmitter-receiver that transmits a digital audio signal within a specified frequency band, and receives the specified frequency band. Chaki modulates a fundamental frequency using Quadrature Phase Shift Keying (QPSK). The QPSK modulated signal is transferred to an infrared emitter for transmission.
U.S. Pat. No. 5,420,640 (Munich, et al.) describes a memory efficient method and apparatus for synchronization detection within a digital data stream over a communication path. The digital data is arranged as a sequence of frames, each frame including a plurality of lines of data. The beginning of each frame is indicated by a frame synchronization word. The beginning of each line is indicated by a horizontal synchronization byte. An encoder, before transmission, interleaves the data. The decoder contains circuitry for locating the horizontal and frame synchronization data and contains circuitry for deinterleaving the digital data. Both the synchronization locating circuitry and the deinterleaving circuitry require access to a memory, but not at the same time. Therefore, a single memory is used with the synchronization recovery circuitry and deinterleaving circuitry alternately addressing the memory. The digital data stream of Munich, et al. pertains to video, audio and other related services of subscriber based television systems.
U.S. Pat. No. 5,745,582 (Shimpuku, et al.) teaches an audio signal transmitting and receiving system which can perform optical transmission of a digital format audio signal with small deterioration of the sound quality over the transmission path. The audio signal transmitting system has circuits to add an error correction signal to a digital audio signal. The digital audio signal with the error correction signal is then encoded and interleaved to generate an audio transmission signal. Repeating a digital control signal, which is to be used for the reproduction of the digital audio transmission signal, generates a continuous signal. A multiplexer combines the audio transmission signal and the continuous signal to generate a multiplexed signal. A modulation circuit then modulates a carrier signal similar to that described above with the multiplexed signal by a predetermined digital modulation method to generate a modulated signal within a predetermined frequency band. The modulated signal is transmitted by an optical transmission signal. A differential type QPSK modulation method creates the modulated signal preferably. Shimpuku, et al. further describes an audio signal receiving circuit for reproducing a digital audio signal and a digital control signal from the optical transmission signal. The audio signal receiving circuit has an optical receiver to convert the optical transmission signal to an electric reception signal. The modulated signal is then reproduced to permit demodulation of the reception signal by a digital demodulation method corresponding to the predetermined digital modulation method to reproduce the multiplexed signal. A separating circuit separates the audio transmission signal and the continuous signal from the multiplexed signal. The audio transmission signal is then deinterleaved and error correction based on the added error correction signal is performed to reproduce the digital audio signal
The digital source 10 is often a compact audio disk (CD), a Moving Picture Expert Group Audio Layer 3 (MP3) data file, a digital audio tape (DAT), a Digital Video Disk (DVD), or a digital satellite receiver (DSR). The format of the digitized samples from a digital source 10 commonly complies with the Sony/Phillips Digital Interface (S/PDIF). International standards that have developed from this standard are the Audio Engineering Society (AES) AES-3, the European Broadcasters Union (EBU) Tech. 3250-E, the Japanese Electronic Industries Alliance (EIAJ) CP-340, and the International Electronic Commission (IEC) IEC60958. While these standards are similar, they are not necessarily identical. However, the data format as shown in FIG. 3 is common for each standard. The allowed sampling frequencies or sampling rates of the audio analog signals to create the digitized samples are the 44.1 kHz for CD and MP3, 48 kHz for DAT and DVD, and 32 kHz for DSR.
Refer now to FIG. 2 to discuss the data format of the S/PDIF family of international standards. A frame consists of two subframes 200 and 205 containing the samples from an A channel or left channel and a B channel or right channel. Each subframe has a synchronizing preamble A SYNC and B SYNC. The synchronizing preamble identifies the contents of the subframe as being either a word containing a sample of the A channel at the beginning of a block 215, the A channel within a block, or the B channel.
The digitized audio samples for channel A and channel B can contain up to 24 bits representing the amplitude of a sample of the analog audio signal. Normally for CD applications, only the 16 bits A8 through A23 are employed to convey the digitized audio samples. The bits AV and BV are the validity indicating if the digitized audio sample is in error. The bits AU and BU are user defined bits which when collected from many samples indicate running time, track number, etc. The bits AC and BC are channel status bits defining such information as emphasis, sampling rate, and copy permit. The bits AP and BP are parity bits for error detection to verify reception of the data samples.
The digitized audio samples are encoded using a commonly known biphase mark or Manchester coding technique. The samples are transferred serially at a rate of 2.8 MHz for a sampling rate of 44.1 kHz, 2 MHz for a sampling rate of 32 kHz, and 3.1 MHz for a sampling rate of 48 kHz.
A block of the digitized audio samples consists of 192 frames concentrated together.
U.S. Pat. No. 5,889,820 (Adams) describes a circuit for SPDIF-AES/EBU digital audio data recovery. The circuit decodes an input signal. The circuit includes a measurement sub-circuit having an input to receive a timing clock signal that is asynchronous with clocking of the input signal. The asynchronous timing clock signal measures the duration of a plurality of pulses received on the input signal in relation to frequency of the timing clock signal. A decode circuit decodes the input signal into digital data. The invention of Adams permits use of all digital components for decoding digital audio data encoding using biphase-mark encoded data according to the S/PDIF or AES/EBU standards.
An object of this invention is to provide a system for transmitting, receiving, recovering, and reproducing digitized samples of analog signals.
Another object of this invention is to conceal unrecoverable digitized samples of analog signals to maintain a level of fidelity in reproducing the digitized samples of the analog signal.
Further, another object of this invention is to transmit the digitized samples of the analog signals such that the probability of interference with the transmission and thus corruption of the digitized samples of the analog signals is minimized by transmitting the digitized samples as bursts shorter period than the time of the analog signals represented by the transmitted digitized samples.
Still further, another object of this invention is to receive the digitized samples of analog signal without synchronizing a receiving clock with a transmitting clock to capture the digitized samples of the analog signals.
Even still further, another object of this invention is to convert the digitized samples of the analog signals from an external source that has various sampling rates to digitized samples of the analog signals having a rate.
Again, another object of this invention is to softly mute the digitized samples of the analog signals when large groups of the digitized samples can neither be recovered nor concealed.
Another object of this invention is to track the long term difference between a transmit clock and a receive clock and to interpolate and decimate any underrun or overrun of the digitized samples of the analog signals within a group of digitized samples of the analog signals.
To accomplish these and other objects, a digital communication system for transmitting and receiving digitized samples of analog signals is comprised of a transmission subsystem for transmitting the digitized samples a communication medium to convey the transmitted digitized samples having various sampling rates, and a receiving system to receive and reproduce the transmitted digitized samples. The transmission subsystem receives the digitized samples having a variable sampling rate from an external signal source and then converts the digitized samples having a variable sampling rate to digitized samples with a fixed rate. The digitized samples have error correction codes generated to allow correction of any errors in the fixed digitized samples that may occur during transmission of the digitized samples. The digitized samples are formatted into groups of interleaved digitized samples with appended error correction codes. A preamble timing signal and a start signal is then appended to the group of interleaved digitized samples to form a transmit frame. A carrier signal is then modulated with the transmit frame and the modulated carrier signal is then transmitted to the communication medium.
The receiving subsystem is connected to the communication medium to receive and recover the modulated carrier signal. The modulated carrier signal is then demodulated to recover the transmit frame and to extract the group of interleaved digitized samples and the error correction codes from the transmit frame. The group of interleaved digitized samples with the error correction codes are checked, and the group of interleaved digitized samples with errors are corrected. If any of the group of interleaved digitized samples are uncorrectable, an estimated sample value of those uncorrectable digitized samples is created by interpolating from adjacent interleaved digitized samples to conceal any effect of the uncorrectable digitized samples. Any of the digitized samples that are not concealable or unrecoverable or are invalid are soft muted. The digitized samples are then transferred to a digital-to-analog converter for restoration of the analog signal.
The transmission subsystem has a sampled data receiver to receive the digitized samples of the analog signals from the external source of the digitized samples of the analog signals. A variable sampling rate converter is connected to the sampled data receiver to convert the digitized samples of the analog signals that are sampled at the one rate of the various sampling rates to digitized samples of the analog signals sampled at a fixed rate. A plurality of the digitized samples of the analog signals is retained in a data buffer. A data buffer controller is connected to the variable sampling rate converter and the data buffer to control the placement and removal of the plurality of digitized samples of the analog signals within the data buffer. An error correction code generator is connected to the data buffer controller to receive multiple digitized samples of the analog signals through the data buffer controller from the data buffer. The error correction code generator generates an error correction word that appended to the multiple digitized samples of the analog signals, and then returns the multiple digitized samples of the analog signals with the appended error correction word through the data buffer controller to the data buffer. The error correction code generator creates a Reed-Solomon error correction code with a the error correction code word that has a data block size of 238 bytes and one control byte and 16 parity bytes. A frame formatter is connected to the data buffer controller to receive an interleaved group of the multiple digitized samples of the analog signals with the appended error correction codes and appends a preamble timing signal and a start signal before the interleaved group of the multiple digitized samples of the analog signals to form a transmit frame. A pulse position modulator is connected to the frame formatter to receive the transmit frame and modulate a carrier signal according to a pulse position modulation with the transmit frame. A burst transmitter is connected between the pulse position modulator and the communication medium to convey a modulated carrier signal to the communication medium. The modulated carrier signal is transmitted as a burst within a short time period to minimize probability of interference on the communication medium.
The communication medium may be either wireless or wired and the modulated carrier signal may be transmitted as light or as Radio Frequency energy. The wired communication media may be either fiberoptic cable, coaxial cable, or two wire twisted pair cable.
The receiving subsystem has a receiver connected to the communication medium to sense and amplify the modulated carrier signal and to recover the transmit frame. A demodulator is connected to the receiver to demodulate the modulated carrier signal and extract the groups of interleaved multiple digitized samples of the analog signals with the appended error correction code. The demodulator is connected to a received data buffer to retain the group of interleaved multiple digitized samples of the analog signals with the appended error correction code. A received data buffer controller is connected to the demodulator and the received data buffer to control transfer of the groups of interleaved multiple digitized samples of the analog signals with the appended error correction code from the demodulator to the received data buffer. An error check and correction circuit is connected to the received data buffer controller to receive one group of the multiple digitized samples of the analog signals with the appended error correction code. The error check and correction circuit checks and corrects any errors that occur in transmission in the one group of the multiple digitized samples of the analog signals and then replaces the corrected group of the multiple digitized samples of the analog signals to the received data buffer. Any non-correctable digitized samples of the multiple digitized samples of the analog signals are identified for concealing. A block recovery circuit is connected to the received data buffer controller to receive and interpolate the non-correctable digitized samples of the analog signals to conceal an effect of the non-correctable digitized samples analog signals. A soft muting circuit is connected to the received data buffer controller to access those groups of the multiple digitized samples that were not correctly received and declared invalid and those of the multiple digitized samples of the analog signals with non-recoverable and non-concealable errors. The soft muting circuit, further, accesses those of the multiple digitized samples of the analog signals that are correct and adjacent to those of the multiple digitized samples of the analog signals that are invalid or with non-correctable and non-concealable errors. The soft muting circuit then applies a smoothing function to the multiple digitized samples of the analog signals to bring those of the multiple digitized samples of the analog signals that are invalid or with the non-correctable and non-concealable error to a null value.
The receiving subsystem has a jitter tracking circuit to compare the block transmission timing signal with a clock signal of the receiver subsystem to determine overrun and underrun of the contents of the group of interleaved multiple digitized samples of the analog signals with the appended error correction code. The block transmission signal indicates a boundary of groups of interleaved multiple digitized samples of the analog signals with the appended error correction codes. The number of words within each group of interleaved multiple digitized samples of the analog signals with the appended error correction codes must contains the correct number of digitized samples of analog signals. An interpolation and decimation circuit is connected to the jitter tracking circuit and the received data buffer controller to generate or eliminate digitized samples of the analog signals if the jitter tracking circuit indicates overrun or underrun of the contents of the group of interleaved multiple digitized samples analog signals.
An interface circuit is connected to the received data buffer controller to translate the digitized samples of the analog signals to a format acceptable by subsequent circuitry.
The digitized samples having variable sampling rates are sampled at sampling rates of 32 kHz, 44.1 kHz, and 48 kHz.
The transmission subsystem may have at least one analog-to-digital converter connected between the external source and the data buffer controller to receive the analog signals and to generate the digitized samples analog signals. The sampling rate of the analog-to-digital converter is approximately 48 kHz. An alternate sampling rate for the analog-to-digital converter is 44.1 kHz.
The interleaved group of the multiple digitized samples is comprised of a plurality of least significant bytes of the even designated digitized samples of the group of multiple digitized samples, a plurality of most significant of the even designated digitized samples, a first command byte, a first plurality of error correction parity bytes, a plurality of least significant bytes of the odd designated digitized samples, a plurality of most significant bytes of the odd designated digitized samples, a second command byte, and a second plurality of error correction parity bytes.
The carrier signal is modulated with a pulse positioned modulation by positioning of a pulse of the carrier signal within a period of the carrier signal according to a binary value of a plurality of bits within the transmit frame. The plurality of bits of this invention is two bits.
The digitized samples are encoded in a non-return to zero (NRZ) format.
The burst transmitter includes an infrared light emitting diode and a diode switching circuit connected between the pulse position modulator and the infrared light emitting diode to activate and deactivate the infrared light emitting diode with the modulated carrier signal.
The receiver comprises a light sensitive diode that receives light radiated from the infrared light emitting diode.
The demodulator demodulates the modulated carrier signal by oversampling the modulated carrier signal to determine an evaluation point of the modulated carrier signal to recover the transmit frame.